Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index beb49bc4ce5bcf5a8cc3b331b63eb600cb20716e..c4518549ad90882d4254ea2145dcc455b2acea39 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -248,7 +248,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   struct InputData {
     uint32_t input_timestamp;
     const int16_t* audio;
-    uint16_t length_per_channel;
+    size_t length_per_channel;
     uint8_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.